Added a flag to AudioCodecSpec to indicate adaptive bitrate support.

webrtc/media/engine/webrtcvoiceengine.cc

Index: webrtc/media/engine/webrtcvoiceengine.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index df93ac28f44db14448bc59ef597d70e5415c71f6..3c15385dbdfd943c597e1abf748b9f95100a75cd 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1159,22 +1159,19 @@ AudioCodecs WebRtcVoiceEngine::CollectRecvCodecs() const {
     rtc::Optional<AudioCodec> opt_codec = mapper.ToAudioCodec(format);
     if (!opt_codec) {
       LOG(LS_ERROR) << "Unable to assign payload type to format: " << format;
-      return false;
-    }
-
-    auto& codec = *opt_codec;
-    if (IsCodec(codec, kOpusCodecName)) {
-      // TODO(ossu): Set this specifically for Opus for now, until we have a
-      // better way of dealing with rtcp-fb parameters.
-      codec.AddFeedbackParam(
-          FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
     }
-    out.push_back(codec);
-    return true;
+    return opt_codec;
   };
 
   for (const auto& spec : specs) {
-    if (map_format(spec.format)) {
+    rtc::Optional<AudioCodec> opt_codec = map_format(spec.format);
+    if (opt_codec) {
+      AudioCodec& codec = *opt_codec;
+      if (spec.supports_network_adaption) {
+        codec.AddFeedbackParam(
+            FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
+      }
+
       if (spec.allow_comfort_noise) {
         // Generate a CN entry if the decoder allows it and we support the
         // clockrate.
@@ -1189,6 +1186,8 @@ AudioCodecs WebRtcVoiceEngine::CollectRecvCodecs() const {
       if (dtmf != generate_dtmf.end()) {
         dtmf->second = true;
       }
+
+      out.push_back(codec);
     }
   }